用Sipp 对Asterisk 进行性能测试的工作笔记-2

xml 文件 . 本文草本出自狼山客人, 保留其相关注释.

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->
<scenario name="Basic Sipstone UAC">

<!--狼山客人：下面这一块表示SIPp发送一个INVITE数据包到SIP server(Wavesplitter的MSP-16)-->
  <send>
    <![CDATA[
   
      INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]              
      Cseq: 1 INVITE                  
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70                
      Subject: Performance Test       
      Content-Type: application/sdp   
      Content-Length: [len]           

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      t=0 0
      c=IN IP[media_ip_type] [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 G726/8000

    ]]>
  </send>

<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个100的数据包-->

  <recv response="100"> optional="true"
  </recv>

<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个200的数据包，如果收到，说明ViVoice公司的VENUS NW800视频电话已经接听了，用户已经提起话筒-->

  <recv response="200">
  </recv>

<!--狼山客人：下面这一块表示SIPp开始通话-->

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[

      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[field0]@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length